Streaming has become increasingly important over the past few years, not only in the fixed Internet, but also in 2.5/3G/LTE mobile networks. In order to support continuous playout of a multimedia stream a transport network has to provide throughput which matches the rate of the encoded content (e.g. guaranteed bit-rate). If this requirement is not fulfilled a buffer underrun or overrun at the client side would occur.
A buffer underrun means that no more data can be played out at the client during re-buffering time (e.g. a frozen image on the screen). Therefore, it is more important to guarantee uninterrupted playout than guaranteeing completely error-free high-quality pictures.
A buffer overrun occurs since mobile devices have more stringent limits to their memory usage than PC clients. An overflow would have an effect equivalent to that of packet loss, as packets causing the overflow are discarded.
Best-effort networks cannot guarantee the same bit-rate during the session lifetime. In mobile networks, wireless links are often characterized by a varying throughput due to the nature of the wireless channel. In addition, different wireless access technologies (e.g. GPRS or HSDPA) pose different maximum limits on the average available bit-rate such that intersystem handovers will result in significantly different link characteristics.
In terms of multicast/broadcast traffic, although it is normally given higher priority than unicast traffic, congestion may still occur if bandwidth reservation is not used. Congestion is not a trivial issue for IPTV (fixed & mobile) services. It is crucial for service providers to closely monitor service quality of such multimedia streaming services and react if there is a problem.
Bit-rate adaptation, or adaptive streaming, is a solution described in M. Westerlund, P. Fröjdh, SDP and RTSP extensions defined for 3GPP Packet-switched Streaming Service and Multimedia Broadcast/Multicast Service, Internet Draft, Ericsson, draft-westerlund-mmusic-3gpp-sdp-rtsp-07, May 8, 2009 for Packet-switched Streaming Service (PSS). PSS is a unicast-based streaming service designed to deliver content at a quality level that requires a rate not exceeding the currently available transmission rate. This is done through varying the bit-rate based on the network conditions. The end-user is always given first the higher bit-rate according to its terminal capability. However, in case of network congestion, the bit-rate will be lowered in a dynamic and seamless way by a Streaming Server. Doing so, the network is less likely to drop packets due to congestion. The playback interruptions can be avoided and the objective playback quality will be at its maximum for the given transmission conditions.
The bit-rate adaptation is server centric, in the meaning that transmission and content bit-rate are controlled by the server, and the client just provides feedback about current buffer status and reception characteristics to the server. This allows the server for adapting the media streams, since it knows how the content is encoded. To support the bit-rate adaptation, the same content needs to be encoded in different bit-rates, but only one channel address is published to the client. The transition between the different bitrates occurs in a transparent way for the terminal which only sees a continuous stream.
Although very efficient for unicast streaming the bit-rate adaptation technique is not applicable to multicast or broadcast streaming because it is not capable of handling quality reports from multiple UEs.
A general assumption of the existing adaptive streaming is that network congestion is the cause of the service problems. This does not have to be true since many other factors, such as UE configuration errors and malfunction of intermediate network nodes, may lead to service problems with the same symptoms. If congestion is not the cause, the bit-rate adaptation may downgrade the media quality unnecessarily. In particular, all of the participants of the same multicast/broadcast session receive the same bit-rate and share the same delivery channel. Down-switch or up-switch of the bit-rate would affect all of the users of the same channel, even those with a very good service quality. Generally, it is unacceptable to downgrade the quality of the session for all users because a fraction of them is affected by the congestion. Because of this, it is not feasible to trigger a bit-rate down-switch (or up-switch) simply based on the observations at one UE (or even multiple UEs). For example, if the mobile TV client at one UE is mis-configured and keeps reporting low throughput, the bit-rate of the channel must not be down-switched to avoid the unfairness.